[asterisk-bugs] [Asterisk 0017956]: atxfer broken on 1.6.2.11
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Sep 22 14:24:57 CDT 2010
The following issue requires your FEEDBACK.
======================================================================
https://issues.asterisk.org/view.php?id=17956
======================================================================
Reported By: ronald_verschaeren
Assigned To: mnicholson
======================================================================
Project: Asterisk
Issue ID: 17956
Category: Features
Reproducibility: always
Severity: major
Priority: normal
Status: feedback
Target Version: 1.6.2.15
Asterisk Version: 1.6.2.11
JIRA: SWP-2167
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
======================================================================
Date Submitted: 2010-09-06 03:14 CDT
Last Modified: 2010-09-22 14:24 CDT
======================================================================
Summary: atxfer broken on 1.6.2.11
Description:
I'm using SIP phones.
- A calls B
- B transfers A to C by pressing ?2, then C's extension (atxfer =>
https://issues.asterisk.org/view.php?id=2 in
features.conf)
- C picks up, talks to B.
- B hangs up.
- both A and C hear silence
afaict, A is not bridged to C. This issue was not present in 1.6.1
======================================================================
Relationships ID Summary
----------------------------------------------------------------------
related to 0017999 Issues with DTMF triggered attended tra...
======================================================================
----------------------------------------------------------------------
(0127293) mnicholson (administrator) - 2010-09-22 14:24
https://issues.asterisk.org/view.php?id=17956#c127293
----------------------------------------------------------------------
I can't reproduce this here. And it looks like the channels are actually
bridged at some point according to this line in your log.
[Sep 6 11:32:13] DEBUG[20871] channel.c: Bridge stops bridging channels
SIP/27-00000020 and Local/90 at default-c5f6;1
Please upload your features.conf, your extensions.conf, and your sip.conf
configuration files.
Issue History
Date Modified Username Field Change
======================================================================
2010-09-22 14:24 mnicholson Note Added: 0127293
2010-09-22 14:24 mnicholson Status acknowledged =>
feedback
======================================================================
More information about the asterisk-bugs
mailing list